Directional microphone systems are designed to sense sound from a particular source such as a desired speaker located in a specified direction while rejecting, filtering out, blocking, or otherwise attenuating sound from other sources such as undesired bystanders or noise located in other directions. To achieve a high degree of directionality, microphones typically include an array of two or microphone sensors or transducers contained in a mechanical enclosure. The enclosure typically includes one or more acoustic ports for receiving sound and additional material for guiding sound from within the beam angle to sensing elements and blocking sound from other directions.
Directional microphones may be beneficially applied to a variety of applications such as conference rooms, home automation, automotive voice commands, personal computers, telephone headsets, personal digital assistants, and the like. These applications typically have one or more desired sources of sound accompanied by one or more noise sources. In such applications, it is desired to increase the signal to noise ratio (SNR) between the desired source and unwanted interferers. Attempts to do so using frequency filtering are largely unsuccessful because the frequencies to be filtered out are typically the same as the desired source, for example, in a telephone headset that seeks to preserve the desired speaker's voice while simultaneously canceling the voices of people other than the speaker such as bystanders. Sound sources other than the desired speaker are referred to herein as interferers.
Because the sound signals from the desired speaker and unwanted interferers are typically emitted from different locations relative to the microphone, the spatial separation between the speaker and interferers can be exploited to separate the desired sound signal from the unwanted interferer sound signal using spatial filters such as a delay-and-sum beamformer or a Griffiths-Jim adaptive beamformer. More specifically, nulls in the directional sensitivity pattern of the microphone array may be used for interference cancellation, while a fixed gain in a known directional location (e.g., corresponding to the desired speaker) may be used to preserve the sound signals emitted by the desired speaker.
For example, FIGS. 1A-1B depict a microphone array 100 having two microphone sensors M1 and M2 positioned along a longitudinal axis 101 and separated by a distance d. A desired speaker (SPKR) is located in the 0 degree (°) direction of the axis 101, and an interferer (INT) is located at an angle θ from the 0° direction of axis 101. Assuming the INT is in the far field, sound waves emitted from INT travel a distance r to M2 and travel a distance r+Δr to M1. Thus, the phase difference in sound signals received at the two sensors M1 and M2, which may be expressed as kΔr=2πΔr/λ, (where λ is the wavelength sound waves), may be used to distinguish between sound signals emitted from the SPKR and from the INT.
A fixed null-steering system such as a well-known beamformer filters the microphone signal produced by sensor M1 and subtracts it from the microphone signal produced by sensor M2 to generate an output signal that suppresses sound signals attributed to INT, thereby creating a fixed sensitivity pattern (also known as polar response pattern). However, in many applications, the location and direction of the interferer (INT) may not be known and/or may change even though the location and direction of the desired speaker SPKR remains constant. In such applications, adaptive filters may be employed to continually modify the system response (e.g., by continuously modifying the polar response pattern) so that the sound processing system steers a “null” in the direction of the interferer. To distinguish between the desired speaker SPKR and the unwanted interferer INT, sound processing systems may employ a combination of fixed beamformers and adaptive filters.
For example, FIG. 2 shows a well-known Griffiths-Jim adaptive beamformer circuit 200 that includes a fixed beamformer and an adaptive filter. Filter circuit 200 is shown to include microphone sensors M1-M2, a delay element 210, subtraction circuits 221-222, summing circuit 223, an adaptive filter 230, and a signal power estimator circuit 240. As depicted in FIG. 2, the speaker SPKR is located along the longitudinal axis of the microphone sensors M1-M2 at a reference angle of 0°. Further, an interferer INT (not shown in FIG. 2) is located at some unknown angle θ relative to the SPKR. In response to sound generated by INT and SPKR, sensor M1 produces a first input signal IN1 and sensor M2 produces a second input signal IN2. IN1 is provided to delay element 210, which is typically a low-pass filter (LPF) that produces a delayed input signal IN1D. Signals IN1D and IN2 are summed at summing circuit 223 to generate a sum signal (SUM) containing signal components of both the SPKR and INT, and signal IN1D is subtracted from IN2 by subtraction circuit 221 to generate a difference signal (DIFF) in which signal components of SPKR are suppressed so that DIFF contains mostly signal components of INT. Thus, sensors M1-M2, delay element 210, and subtraction circuit 221 together form a fixed beamformer that suppresses SPKR from DIFF in a well-known manner, for example, by setting the filter coefficients of delay element 210 to suitable values according to the distance between sensors M1-M2 and the direction of SPKR (which is at 0° in FIG. 2).
The difference signal is provided as an input signal to adaptive filter 230, which includes an output to generate a filtered difference output signal FD and includes a control terminal to receive a tuning signal from signal power estimator (SPE) 240. The filtered difference signal FD is subtracted from SUM in subtraction circuit 222 to generate an output signal OUT that dynamically preserves sound components of SPKR while suppressing sound components of INT over a range of changing directions for INT.
As known in the art, SPE circuit 240 estimates the signal power of the output signal OUT, and in response thereto generates a tuning signal (TN) that is used to continuously tune the adaptive filter 230. Although not shown for simplicity, for some applications, the SPE circuit generates the tuning signal TN for the adaptive filter 230 in response to both the output signal OUT and the difference signal (DIFF). Adaptive filter 230, which is typically a finite impulse response (FIR) filter, is continuously tuned in response to TN to suppress the dominant source components in DIFF so that INT sound components are suppressed from its output signal FD. More specifically, the polar response pattern of adaptive filter 230 is continuously modified to continuously steer the null in the direction of INT to minimize the sound energy attributed to INT from the filtered difference signal FD.
It is important to note that adaptive beamformers of type shown in FIG. 2 are implemented using digital circuitry, for example, because FIR filters operate in the digital domain.
Thus, when the filtered difference signal FD is subtracted from the sum signal SUM at subtraction circuit 222, the resultant output signal is a directionally sensitive signal in which the INT components are suppressed and the SPKR components are preserved. For example, if the sum signal SUM is represented as a SPKR component S plus an INT component INTSUM and the filtered difference signal FD represents the estimate of ISUM the output signal OUT=S+INTSUM−FD≈S, and the transfer function of the adaptive filter is H(ω)=INTSUM/FD.
Although effective in providing a directional sensitivity pattern that can dynamically steer a null in the direction of INT, the adaptive filter employed by systems such the Griffiths-Jim circuit 200 requires a complicated algorithm to continuously steer the null in the direction of the interferer INT. In addition, the adaptive filter itself is typically a very complex circuit requiring numerous cascaded filtering stages and various adjustable tap delay lines, which not only consumes a large circuit area but also may be difficult to design and implement.
Applicant has developed a response select null steering circuit that includes a beamformer, a summing circuit, a plurality of separate filtering circuits, and a selection circuit. In response to input signals generated by microphone sensors receiving sound signals from a desired speaker and an unwanted interferer, the summing circuit generates a sum signal containing signal components of both the speaker and the interferer. The beamformer generates a difference signal that suppresses signal components of the desired speaker so that the difference signal contains primarily only the signal components of the interferer. Each filtering circuit includes a fixed filter and a subtraction circuit that together provide a different polar response pattern that exhibits a null in a unique direction relative to the desired speaker. In this manner, each filtering circuit may be individually configured to suppress sound signals from an interferer located in a direction associated with the null in the corresponding polar response pattern of the filter. The selection circuit receives the output signals from the various filtering circuits and selects the output signal that has the least amount of signal energy, where the output signal having the least signal energy achieves the best suppression of the unwanted interferer.
However, existing module testers are not sufficient to properly test such response select null steering circuits. As a result, a new module tester is needed that can properly test and calibrate circuits such as the response select null steering circuits described herein.
Like reference numerals refer to corresponding parts throughout the drawing figures.